1. Field of the Invention
The present invention relates to communication systems and session establishment methods, and more particularly, to a communication system for performing communication with a session established between terminals via a network and a session establishment method for establishing a session between terminals via a network.
2. Description of the Related Art
In recent years, IP (Internet Protocol) telephony services (Internet telephony services) permitting voice telephony over an IP network have become available. IP telephony can be provided at low call rates, compared with the existing telephony, and thus is expected to come into wide use with the advance in the network technologies in future.
Unlike ordinary telephone networks in which the connection with a called party is established prior to communication, IP networks generally adopt a connectionless communication form in which information is transmitted directly to the called party without the connection being ascertained.
Accordingly, to implement telephone communication in connectionless communication environments such as IP networks, it is necessary to employ a protocol (connection-type protocol) which allows communication to be initiated after the connection with a called party is ascertained, and a protocol called SIP (Session Initiation Protocol), for example, is used in IP telephony services.
SIP is a connection-type signaling protocol which, using IP packets, initiates and terminates voice calls as well as multimedia communication of, for example, the video of videophones between terminals connected to IP networks.
Electronic mail, for example, is one-way communication of information, and therefore, when email is transmitted, the transmitting-side application and the receiving-side application need not be associated with each other (session need not be established).
In the case of IP telephone calls, on the other hand, it is necessary that the calling party and the called party should both be on the phone at the same time. Thus, to ascertain that both parties are on the phone, for example, the IP telephony applications of the calling and called parties need to be associated with each other through a series of steps: “notification of dialing by the calling side”→“ringing of the called side”→“answering by the called side”.
Such “association (connective relation)” between the applications over the Internet is called session, and call control including the initiation of a session is taken care of by SIP.
FIG. 14 shows a network configuration adapted for IP telephony. An IP network 100 includes a SIP server 101, and client terminals 102 and 103 are connected to the IP network 100. The SIP server 101 is a device for controlling the connection/disconnection of a session between the client terminals 102 and 103.
The client terminals 102 and 103, which make use of IP telephony service, are respectively assigned unique SIP-URIs (SIP-Uniform Resource Identifiers) as their telephone numbers (communication resource identifiers) to be uniquely identified on the network, and a session is established based on the SIP-URIs.
When the client terminal 102 as an originating side makes an IP telephone call to the client terminal 103 as a receiving side, first, the originating-side client terminal 102 connects with the receiving-side client terminal 103 by using SIP (signaling protocol).
At this time, the SIP server 101 receives a session establishment request from the originating-side client terminal 102, whereupon the SIP server identifies the receiving-side client terminal 103 from the telephone number (SIP-URI) specified as a target of session request by the originating-side client terminal 102 and transfers the session establishment request from the originating-side client terminal 102 to the receiving-side client terminal 103.
After a session is established, voice data and the like are exchanged by means of a streaming protocol such as RTP (Real-time Transport Protocol) and RTCP (Real-time Transport Control Protocol).
SIP itself provides only the basic call control function such as the initiation, change, and disconnection of sessions. Accordingly, to provide an encryption function for secure communication via sessions, SSL (Secure Socket Layer) or TLS (Transport Layer Security), for example, is used as in HTTP (Hyper-Text Transfer Protocol). Also, as a protocol for describing sessions, SDP (Session Description Protocol) is used, and to provide a transfer function for data requiring simultaneity such as voice and video, a transport protocol such as RTP or RTCP is used.
Thus, SIP is a protocol for performing only the basic session control and is used in combination with a plurality of other protocols so that the functions thereof may complement one another to realize communication between clients.
SIP imposes no restrictions on the type of data to be exchanged via sessions and, therefore, has a wide range of application. For example, IP telephony can be implemented by exchanging voice between client applications via sessions controlled by SIP, and video telephony can be implemented by exchanging voice and video via such sessions.
Since SIP is simple, highly expansible and is designed so as to make good use of the existing protocols, a variety of uses are considered as possible applications requiring real-time functionality such as video telephony, video chat and videoconferencing, besides IP telephony. SIP is therefore attracting attention as a potential major protocol for realizing real-time communication over IP networks.
Meanwhile, when information is communicated via networks, it is necessary that personal information should be properly managed. As such information management techniques, a technique is known in which set values relating to permission of personal information disclosure to third parties are hierarchically managed in accordance with disclosure levels (e.g., Unexamined Japanese Patent Publication No. 2005-38393 (paragraph nos. [0024] to [0042], FIG. 1)).
When IP telephony communication is established between clients, the originating-side client can establish communication without notifying the receiving-side client of the actual telephone number (SIP-URI), which is personal information on the originating-side client, by using the originator number blocking setting as provided by RFC3325 (Private Extensions to SIP for Asserted Identity within Trusted Networks) or draft-ietf-sip-privacy-02 (SIP Extensions for Caller Identity and Privacy), for example.
However, the telephone number of the receiving-side client as the called party must always be specified at the time of requesting session establishment, and it is therefore necessary that the originating-side client should know the telephone number of the receiving-side client. Conventionally, therefore, users of IP telephony services are eventually compelled to disclose their telephone numbers to each other. However, some users may leak other users' telephone numbers, creating the possibility of malicious attacks. Accordingly, users must refrain from thoughtlessly disclosing their telephone numbers to other people than acquaintances.
Also, if a user's telephone number is leaked, the telephone number needs to be changed, and this forces the user to take the trouble to notify all of the clients who already know the leaked telephone number that the current telephone number will be changed.
Meanwhile, as illegal use prevention services for electronic mail, there have been developed anti-spam mail services enabling setting of conditional mail addresses (e.g., in Japan, “privango” mail service), wherein a temporary mail address linked to the user's real mail address is issued by an issuing server and the conditions for using the temporary mail address are set such that, for example, mail from only a specified mail address is accepted for only a limited term of use.
Such mail services are used in the following manner: Usually, when purchasing goods via the Internet, for example, the user needs to register his/her mail address on the corresponding Web site. In such cases, the user registers, on the Web site, the conditional mail address (temporary mail address issued by the issuing server) with respect to which the user's desired term of use and the user's designated mail address are specified, instead of the real mail address.
Thus, even if the mail address registered on the Web site is leaked because of illegal act or the like, all mail directed to that address is blocked by the mail server after a lapse of the set term of use. Also, since the sender's mail address is specified, all mail from addresses other than the specified sender's mail address is blocked by the mail server.
The use of such services makes it unnecessary for the user to disclose his/her real mail address to other people so that the user can make his/her mail address (conditional mail address) known without anxiety even to a person he/she meets for the first time.
However, the conditional mail address issued by the issuing center is composed of a meaningless random character string, which is difficult to remember. Accordingly, the user may feel reluctant to use the mail address or feel it troublesome to tell the mail address to other people. Further, if the issued mail addresses leak out, mail directed to the leaked mail addresses can be blocked by the mail server, but the load on the issuing center or the network increases because increased mail is sent to the center.
According to the aforementioned conventional technique (Unexamined Japanese Patent Publication No. 2005-38393), personal information is hierarchically managed by a database in accordance with the disclosure levels, and thus, the technique is not suited for the protection of real mail addresses etc. which are very likely to leak out during the call control.